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Documentation Index

Fetch the complete documentation index at: https://www.bolna.ai/docs/llms.txt

Use this file to discover all available pages before exploring further.

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How outbound calls work with BYOT

Outbound calls from your SIP trunk are placed via the standard Bolna call API. Bolna looks up the from_number, resolves the associated trunk and gateway, and places the call via Asterisk’s PJSIP channel driver using your trunk’s credentials.

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Prerequisites

Before making outbound calls, ensure:
  1. You have created a SIP trunk on Bolna
  2. You have added your phone numbers to the trunk
  3. is_active is true on the trunk
  4. You have a Bolna agent created

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Step 1. Configure the agent’s telephony provider

Set telephony_provider to "sip-trunk" in your agent’s configuration. This tells Bolna to route outbound calls through your SIP trunk and use ulaw audio encoding. 
curl -X PATCH https://api.bolna.ai/v2/agent/{agent_id} \
  -H "Authorization: Bearer <token>" \
  -H "Content-Type: application/json" \
  -d '{
    "agent_config": {
      "telephony_provider": "sip-trunk"
    }
  }'

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Step 2. Place an outbound call

Use the standard call API, specifying the from_number as a DID registered on your trunk: 
curl -X POST https://api.bolna.ai/call \
  -H "Authorization: Bearer <token>" \
  -H "Content-Type: application/json" \
  -d '{
    "agent_id": "agt-xxxxxxxx-xxxx-xxxx-xxxx-xxxxxxxxxxxx",
    "recipient": {
      "phone_number": "+918800001234",
      "name": "Rahul Sharma"
    },
    "from_number": "+919876543210"
  }'
The from_number must be a phone number you have registered on your SIP trunk. Bolna will use the trunk’s gateway and credentials to place the call.

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Troubleshooting outbound calls

If outbound calls are failing or not being placed:
  1. Confirm is_active is true on the trunk
  2. Confirm the from_number used in the call request is registered on the trunk
  3. Confirm the gateway address and credentials are correct — check your provider portal for authentication errors
  4. For ip-based trunks, ensure 13.200.45.61 is in your provider’s IP whitelist
  5. Confirm outbound_leading_plus_enabled matches what your carrier expects (some carriers reject +, others require it)

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No audio / call connects but no voice

A common cause is an SRTP mismatch with your carrier. If the trunk has media_encryption="sdes" but the carrier has SRTP disabled (or vice versa), the SIP call sets up but the media path never establishes. Either align both sides on SDES, or set media_encryption="no" and use UDP/TCP transport. See SIP Trunk setup for details.

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Audio quality issues / one-way audio

  • Ensure rtp_symmetric and force_rport are both true (the defaults)
  • Confirm your SIP provider’s RTP IP ranges are reachable from 13.200.45.61
  • Verify allow includes ulaw
  • Avoid enabling direct_media unless explicitly advised by Bolna support

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Next steps